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24 Dec 13 Testing dial tone and Outbound calling from Cisco Router and FXO Card

Dial tone and Outbound calling – Cisco Router/FXO Card

Recently I had a need to test and determine if there was a dial tone on the Cisco FXO Card installed into a Cisco 2911 router. I knew that telco terminated the pots lines on the 66 block but I did not know if they were plugged into the Cisco FXO card. I did not have any one onsite to give me a visual confirmation. So I found out about couple of cool debug commands to accomplish this:

debug vpm signal
debug vpm all

Here are the steps I took:

  • First I used a test Cisco IP phoneon my desk and set it up with this offices extension in Cisco Call Manager
  • Next I setup the debug commands debug vpm all on the router with term mon
  • Next I tried to dial out from my test phone and since the Cisco Call Manager was setup to use the FXO card for the calling, it tried to dial out using one of the FXO ports
  • Now take a look at the debug message from the Cisco Router

#htsp_allocate_if —

Nov 27 10:31:46.468: HTSP endpoint_info=aaln/S0/SU0/0, type=2, under_specified=0,
service_type=2htsp_allocate_if: MATCH!

Nov 27 10:31:46.472: htsp_timer_stop3 htsp_setup_req
Nov 27 10:31:46.472: Orig called num:16152324144
Nov 27 10:31:46.472: htsp_process_event: [0/0/0, FXOLS_ONHOOK, E_HTSP_SETUP_REQ]fxols_onhook_setup
Nov 27 10:31:46.472: [0/0/0] set signal state = 0xC timestamp = 0
Nov 27 10:31:46.472: dsp_set_sig_state: [0/0/0] packet_len=12 channel_id=128 packet_id=39 state=0xC timestamp=0x0
Nov 27 10:31:46.472: TGRM: reg_invoke_tgrm_call_update(0, 0, 0, 65535, 1, TGRM_CALL_BUSY, TGRM_CALL_VOICE, TGRM_DIRECTION_OUT)
Nov 27 10:31:46.472: htsp_timer – 1300 msec
Nov 27 10:31:46.728: htsp_process_event: [0/0/0, FXOLS_WAIT_DIAL_TONE, E_DSP_SIG_1100]fxols_power_denial_detected
Nov 27 10:31:46.728: htsp_timer2 – 1000 msec
Nov 27 10:31:46.728: htsp_timer_stop
Nov 27 10:31:47.728: htsp_process_event: [0/0/0, FXOLS_WAIT_DIAL_TONE, E_HTSP_EVENT_TIMER2]fxols_power_den_disc
Nov 27 10:31:47.728: htsp_timer_stop
Nov 27 10:31:47.728: htsp_timer_stop2
Nov 27 10:31:47.728: [0/0/0] set signal state = 0x4 timestamp = 0
Nov 27 10:31:47.728: dsp_set_sig_state: [0/0/0] packet_len=12 channel_id=128 packet_id=39 state=0x4 timestamp=0x0
Nov 27 10:31:47.728: mars_flex_dsprm_current_codec_comp:DSP:0 FLEX Complexity Codec htsp_release_req: cause 34, no_onhook 0
Nov 27 10:31:47.728: htsp_process_event: [0/0/0, FXOLS_ONHOOK, E_HTSP_RELEASE_REQ]fxols_onhook_release
Nov 27 10:31:47.728: TGRM: reg_invoke_tgrm_call_update(0, 0, 0, 65535, 1, TGRM_CALL_IDLE, TGRM_CALL_VOICE, TGRM_DIRECTION_OUT)
Nov 27 10:31:47.728: flex_dsprm_close_cleanuphtsp_allocate_if —

Nov 27 10:31:47.780: HTSP endpoint_info=aaln/S0/SU0/1, type=2, under_specified=0,
service_type=2htsp_allocate_if: MATCH!

If you look at lines above you will notice that it clearly did not get a dial tone and hence my called failed. With this information I was able to verify if someone connected the cables from the 66 block to the Cisco FXO Card. I could have also used this same technique to see if there was dial tone on the pots lines on the 66 block. In case there is no butt set is available

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